1. Field of the Invention
The present invention relates to a speech compression coding device which is applied to a phone answering system, a voice response system, voice mail and so forth. In detail, the present invention relates to a speech compression coding device which receives an analog speech waveform, converts it into a digital speech waveform, codes the digital speech waveform with a predetermined coding method and thus compresses the amount of data representing the speech.
2. Description of the Related Art
Recently, there has been a need for enlargement of channel capacity of vehicular communications such as that using mobile telephone systems and storage and transmission of very large amounts of information in multimedia communication. Accordingly, practical low bit-rate speech coding is needed
Further, as an additional function of a facsimile modem, development of a speech coding method for a phone answering system is needed.
Currently, a CELP (Code Excited Linear Prediction) coding system has been mainly used, as a low bit-rate speech compression coding system of not more than 10 kbps. The CELP coding system is a coding system based on speech AR (Auto-Regressive) models based on linear prediction.
Specifically, on a coding side, a speech signal is divided into frames or sub-frames. Then, for each unit, LPC (Linear Prediction Coding) coefficients which represent the spectrum envelope, a pitch lag which represents pitch elements, stochastic elements and gains are extracted. Each extracted information is coded and stored or transmitted.
On a decoding side, each coded information is decoded, an excitation vector signal is generated as a result of adding the pitch elements to the stochastic elements. The excitation vector signal passes through a linear prediction synthesis filter which is formed using the LPC coefficients. Thus, synthetic speech is obtained.
However, in the CELP coding system of the prior art, although good speech can be obtained at a low bit rate of 10 kbps, the amount of calculation required for extracting and coding each parameter is large.
In particular, with regard to extracting and coding of pitch lag and extracting and coding of stochastic elements, it is necessary to generate synthetic speech by causing an excitation vector signal to pass through a linear prediction synthesis filter and compare the synthetic speech with the original speech. However, because a large amount of calculation is necessary for the filter operation, it is unpractical to cause all excitation vector signals to pass through the filter.
Further, in the CELP coding system in the prior art, a codebook for a second error signal is provided. A second error signal is synthesized from each code vector of the codebook and the spectrum envelope. Then, the synthesized second error signal is compared with the second error signal obtained from an input signal. The code vector by which distortion of the synthesized second error signal from the second error signal of the input signal is at a minimum is selected. Thus, extracting and coding is performed. However, in this method, a large amount of calculation for the codebook search and a large storage capacity of memory for storing the codebook are needed.
As prior art for reducing the amount of calculation in the CELP coding system, a pre-selection method has been proposed. The method uses a parameter by which an approximate comparison with original speech can be conducted without performing a filter operation so that the number of candidate code vectors is reduced. Then, the filter operation is performed on the reduced number of candidate code vectors, and thus, one of the code vectors is selected.
Further, generally speaking, a random codebook includes the number of stochastic vectors for a given number of bits. A method for reducing an amount of calculation by devising the arrangement has been proposed. Specifically, for example, in the VSELP (Vector Sum Excited Linear Prediction) coding system, the number of stochastic vectors which is the same as the number of bits are provided. Then, adding and/or subtracting these stochastic vectors with each other, various stochastic vectors can be obtained.
However, a practical low bit-rate speech coding is needed, methods for reducing the amount of calculation are needed other than the methods in the prior art of reducing the amount of calculations such as a preliminary selecting method, a VSELP coding method and so forth.